Applicant""s invention relates to electrical telecommunication, and more particularly to wireless communication systems, such as cellular and satellite radio systems, for various modes of operation (analog, digital, dual mode, etc.), and access techniques such as frequency division multiple access (FDMA), time divisional multiple access (TDMA), code divisional multiple access (CDMA), hybrid FDMA/TDMA/CDMA, for example. More specifically, this invention relates to methods and systems which detect multiple information streams which are transmitted as a composite signal in a manner intended to improve the detection of the individual streams.
In North America, digital communication and multiple access techniques such as TDMA are currently provided by a digital cellular radiotelephone system called the digital advanced mobile phone service (D-AMPS), some of the characteristics of which are specified in the interim standard TIA/EIA/IS-54-B, xe2x80x9cDual-Mode Mobile Station-Base Station Compatibility Standardxe2x80x9d, published by the Telecommunications Industry Association and Electronic Industries Association (TIA/EIA) and some of which are specified by the later interim standard IS-136 (which describes, among other things a digital control channel) which standards are expressly incorporated herein by reference. Because of a large existing consumer base of equipment operating only in the analog domain with frequency-division multiple access (FDMA), TIA/EIA/IS-54-B is a dualmode (analog and digital) standard, providing for analog compatibility together with digital communication capability.
In a TDMA cellular radiotelephone system, each radio channel is divided into a series of time slots, each of which contains a burst of information from a data source, e.g., a digitally encoded portion of a voice conversation. The time slots are grouped into successive TDMA frames having a predetermined duration. The number of time slots in each TDMA frame is related to the number of different users that can simultaneously share the radio channel. If each slot in a TDMA frame is assigned to a different user, the duration of a TDMA frame is the minimum amount of time between successive time slots assigned to the same user.
The successive time slots assigned to the same user, which are usually not consecutive time slots on the radio carrier, constitute the user""s digital traffic channel (DTC), which may be considered a logical channel assigned to the user. In only one of many possible embodiments of a TDMA system as described above, the TIA/EIA/IS-54-B, IS-136 standards provided that each TDMA frame consists of six consecutive time slots and has a duration of 40 milliseconds (msec) as illustrated in FIG. 1. Thus, each radio channel can carry from three to six DTCs (e.g., three to six telephone conversations), depending on the source rates of the speech coder/decoders (codecs) used to digitally encode the conversations.
Such speech codecs can operate at either full-rate or half-rate. A full-rate DTC requires twice as many time slots in a given time period as a half-rate DTC, and in TIA/EIA/IS-54-B and IS-136, each full-rate DTC uses two slots of each TDMA frame, i.e., the first and fourth, second and fifth, or third and sixth of a TDMA frame""s six slots. Each half-rate DTC uses one time slot of each TDMA frame. During each DTC time slot, as seen in FIG. 2, 324 bits are transmitted, of which the major portion, 260 bits, is due to the speech output of the codec, including bits due to error correction coding of the speech output, and the remaining bits are used for guard times and overhead signalling for purposes such as synchronization.
Once information has been output from the speech codec, it is then processed for transmission on a radio carrier. This processing can be generalized as illustrated in the upper branch of FIG. 3. Therein, channel coding 30 and interleaving 32 are provided to protect against channel errors which corrupt the information as it is transmitted over the radio channel 36. Channel coding, e.g., block coding or convolutional coding, adds redundancy to the information stream which can be used to identify and correct errors which occur during transmission of the information over the radio channel. Bit errors which occur due to transmission over the radio channel frequently occur in bursts. However, certain types of channel coding are most effective at correcting single bit errors and are less effective at correcting long strings of erroneously received bits. Accordingly, interleaving is used to separate consecutive information bits and transmit them in a non-consecutive manner. In this way, burst errors are effectively spread out so that when the received information is de-interleaved, the channel coding is more likely to be able to correct the errors which occurred during transmission.
Different systems use different types of channel coding and interleaving. For example, systems designed in accordance with the IS-136 standard system described above can provide for channel coding and interleaving, according to one vocoder described therein, as illustrated in FIG. 4. Therein, output of the speech coder 40 is separated into class 1 and class 2 bits, class 1 bits being more important than class 2 bits in terms of the perceived signal quality upon reproduction and, therefore, being more heavily protected against errors. In fact, to further protect the 12 most perceptually significant class 1 bits, a 7 bit cyclic redundancy check (CRC) is computed over those 12 bits at block 42 and added to the string of bits to be convolutionally encoded at block 44. In convolutional encoding, an output, coded bit depends not only on the bit value of a most recently input bit, the bit values of preceding bits, which provides a form of memory that can be used to detect errors in the received signal stream. The rate of convolutional coding, in this example xc2xd, denotes the amount of redundancy providedxe2x80x94in this case for every input information bit, two coded bits are produced. The coded class 1 bits and the uncoded class 2 bits are then ciphered (block 46) and interleaved (block 48) over two time slots as shown in FIG. 5. Thus, bits from each of two speech frames are transmitted in each time slot of D-AMPS systems to spread out burst errors as described above.
Returning to FIG. 3, the output of interleaver 32 is sent to modulator 34 wherein the data is modulated onto the radio frequency carrier. In the D-AMPS example described above, the particular modulation which is currently used is xcfx80/4 shifted, differentially encoded quadrature phase shift keying (DQPSK). In this scheme, as will be appreciated by those skilled in the art, information modulation is achieved by relative changes in phase of the modulating waveform. Grey coding is used in the constellation mapping (described below) of symbols to di-bits so that adjacent signal changes differ by only one bit. In this way, noise errors which result in the erroneous selection of a symbol associated with an adjacent phase create only a single bit error. Once the information is modulated, some post-processing (e.g., filtering and amplification) may be performed and the information is then transmitted over the radio channel.
For completeness, FIG. 3 also indicates functional blocks associated with a receiver, e.g., in a mobile station, that process the received signal. Therein, demodulator 38, de-interleaver 41 and decoder 43 effectively reverse the processes performed by modulator 34, interleaver 32 and channel encoder 30, respectively. Those skilled in the art will be familiar with the operation of these devices and, therefore, they are not further described herein. An optional equalizer 39 (or RAKE receiver, e.g., for a DS-CDMA system) may also be included in the signal processing path. This device handles the effects of signal reflections which occur during transmission of the information over the radio channel by, e.g., creating a model of the channel and attempting to determine the most likely transmitted sequence in view of the various echoes received during some reception interval.
As mentioned earlier, information in IS-136, as well as many other systems, can be transmitted at full or half rate. Half rate communications provide an opportunity for additional capacity in terms of the number of connections, since each frame supports, for example, six channels instead of three. However, the two-slot interleaving described above is applicable only to full rate transmissions, since half rate transmissions only use one time slot per frame. Thus, current implementations of half rate communications in IS-136 systems provide for no inter-slot interleaving and, accordingly, suffer from burst error correction problems.
Accordingly, it would be desirable to identify solutions to provide for half rate communications which overcome these problems and drawbacks. More generally, it would be desirable to provide systems and methods which consider multi-user detection wherein multiple users or sources transmit information in an overlaid or interleaved manner.
These and other drawbacks and limitations of conventional methods and systems for communicating information are overcome according to the present invention, wherein Applicants present techniques and systems for multiplexing two users or sources into each time slot at half rate. This multiplexing technique provides, effectively, two sub-channels of information, which in turn provides an opportunity to improve decoding/demodulating of each sub-channel by re-encoding/remodulating information after successful decoding/demodulating of one sub-channel. Various modulation constellations are described which take advantage of the fact that certain bits in each symbol may be known.